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how do YOU read the RS spl meter?


Scp53

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I have not had time to digest the attachments. However, I've been at home from minor surgery and so had time to run the Rives Audio test CD and the analog RS meter test. I'm not going to post the results because they are so particular to my total room + system response as to be meaningless to others. That having been said, I think that using any meter with sine wave tones, as on the Rives audio CD, is a very gross measurement. The standing wave problems, especially above 1 KHz are amazing in any normal room. Also, the acoustic shadow of my body kept influencing readings, even though I was standing to the side of the meter. I therefore had to let the meter settle as I looked at it. The test tones are rather brief and I had to operate with the CD remote in one hand and the pencil in the other. The meter was on a tripod. BTW, this first test was in 7.1 stereo mode, with "room correction" EQ and all speakers (4 CW IIs and a homemade center) on. I stopped the test at 8 Khz to avoid tweeter damage. My point is, the RS meter is really more of an environmental tool (i.e., for gross levels of music and noise); if you want to try to measure "frequency response" you need two things: education in the process and a true cal'd mic and PC-based analyzer, which now can be had for about $500.

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Or you could go with tools (TDS &/or MLS) that accurately isolate the direct and reflected components and allow you to focus on all aspects of each independently. Then you can identify what characteristics of each require adjustment, you can selectively 'tune' them, and you can verify that what you did is indeed what you wanted to do!

The big difference in the two methods being that with MLS you will go home stir crazy from listening to whoops (measurement sweeps) all day long and with TDS you can put on whatever music you want to listen to and run the sweeps in the background and go home with a modicum of sanity still remaining!

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I thank you all for the informative responses. It may be a waste of time (per dragon fry), but I will enjoy the effort of trying to balance my system with the equalizer. This is my only choice as the area is cannot be easily modified (it is our living room) and I have the "wife factor" to consider. As one poster noted, an EQ can help some overly bright speakers and I had previously noticed that the EQ helped when I was running the Onkyo TX777 with the Klipsch. The new Marantz will allow to tweak each channel individually rather that just the left and right fronts as the Onkyo did.

One more question, does it matter if the RS meter is on it's side when measuring? It would make it easier, when mounted on the tripod, to be able to read from one side if the scale face was visible from that angle rather than trying to look down on the meter from above and behind.

Thanks again

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Nope! The side is fine.

And don't worry too much about the 'absolute' value of the measurement, as you are more concerned with the relative values of each measurement - in other words, how each measurement compares with the other. As long as all he conditions are constant, you will get a pretty good idea.

And by all means have fun! I mean, how much can your really goof things up!? And I can't imagine you doing anthing serious enough to hurt yourself or anyone else!2.gif9.gif

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On 6/15/2005 9:04:56 PM N wrote:

Conclusion : EQ cant fix problems in the time domain,the only fix is physical placement?

Picture host - imageshack.us

PM sent dragonfyr

Nice articles,will read after ive done some CISCO
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More like: EQ can't fix acoustical problems because they change throughout the time domain. The correct fix for an acoustical problem is to fix the acoustics 2.gif

(There are other factors that exist in the time domain, but acoustics is the biggest culprit)

Btw, EQ is a totally valid tool for correcting any issues in the signal chain all the way up to the direct sound of the speaker provided you stay within the limitations of the equipment. In other words, EQ works when you're not dealing with the time domain.

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Can't the eq also be used to adjust for shortcomings, both real and perceived, from the loudspeaker itself?

I thought the discussion was concerning that eq could not adjust for room effects/acoustics. I do understand the timing and room node debate. Good thread guys.

Michael

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On 6/16/2005 9:11:33 AM colterphoto1 wrote:

Can't the eq also be used to adjust for shortcomings, both real and perceived, from the loudspeaker itself?

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Yes, in a way the loudspeaker itself is mostly an electrical device anyway. A lot of the cool crossover designs out there aren't just making the two speakers the same volume at the crossover...sometimes they tweak the slopes and even use different crossover points for the different drivers so as to achieve a flat frequency response in the frequency ranges that the crossovers are active. (Think if your woofer has a slight bump in the response around where you need the crossover, crossing it over steeper and perhaps a bit lower will help to reduce things like this).

However, EQ doesn't really fix issues due to driver position within the loudspeaker. For example, time-aligning the drivers isn't possible in the electrical realm...even if you put delays on the individual units themselves. The reason you can't do it is because moving the driver forward and backward phyiscally changes the polar response in comparison to the other drivers as well (whereas just moving it back and forth electrically does not). There's an article over at prosoundweb that talks about this (with fancy pics too). Anyways, whether or not you consider this to be "acsoutics" is just a matter of terminology...the fact that you're dealing with differences over time means you need to treat it that way.

Btw, I just wanted to add that even though EQ is not the correct tool to solve acoustical issues, I feel you can still improve the sound with an EQ depending on what sacrifices you're willing to make. For example, room nodes are rarely 100% destructive so there's always some room to boost that frequency and notice somewhat of an improvement. The reason it sometimes sounds wierd when we do too much of a boost is because it is also boosting the level of the direct sound, which is arriving to our ears first. The reflections from the room causing the node arrive just slightly later. The end result is too much boost causes a louder direct sound, which then gets subtracted back to the original level (as when there was no boost because increasing the volume will also increase the amount of destructive intereference at the same time). (Yes I know I'm over simplifying too). I know the auto EQ thing at colter's place really made a positive difference on everything we were listening to and you really gotta trust your ears. In fact, I've never been in a position where a little EQ couldn't help the sound...whether it be at home listening to music or mixing live for bands. When on the road you can't exactly change the acoustics of the venue so relying on very careful speaker placement and EQ is the compromise.

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DrWho mentions some good points.

My only qualification is regarding the 'but' to the notion of an improvement in room acoustics.* With EQ, we simply move some of the issues ('nulls') around slightly - as is commonly done in large venue live sound - but as few have really had the opportunity to examine the frequency dependent polar response of a large scale concert system, let me assure you that there is plenty of 'weirdness' happening! (And ask any FOH engineer, the nulls of course ALWAYS correspond to ONLY the aisles! And I have some prime beachfront property for sale on Mars too! 2.gif )

Oh, and just to clarify...the comb filtering and the polar response are corollaries of one another. They display aspects of the same phenomena from two different perspectives, each with advantages and disadvantages - meaning that each display presents various aspects in a manner that is not as apparent in the other view. They are NOT separate 'things'.

One more thing that folks should realize... time and frequency domain issues exist not only at the room-speaker interaction level, but also at the level of component and enclosures in speakers as well! So it is rather like a multi-level puzzle much like the matryoshka, or Russian nested dolls!

So we never seem to get away from the topic, although some specific interactions are more critical then others.

It is rather like a micro and macro examination of system interaction, and they 'overlap' in various ways!

And if anyone is interested and has two very small speakers available, let me suggest an experiment/demonstration that is as impressive as any (surpassed ONLY by the direct AB comparison of a Bessel array with all of the other array topologies) I have heard. And, if you have any kids, it is a basis for a dramatic and winning science fair project as well! (Let me know and I would be glad to provide additional info and assistance).

---------------------------------------------------

Since they are easy to handle, you will need two 3" cube speakers **, with the ability to feed each speaker with the same identical mono single frequency tone. (Be careful with driver heating!)

First note the frequency and polar response of one of the speakers.

Now stack them vertically, one atop the other. Feed a split identical (mono) signal to both speakers ***.

Note the change in frequency response and the polar pattern!

Now very gently and slowly move the top speaker back very slowly. As you do, the polar response in the room will create a 'steered' pattern as if a CD horn was revolved about its axis - much as if you had a revolving 'air raid/Civil Defense" horn or a single ended Leslie cabinet in the speaker position. The sound coverage pattern will change in a manner analogous to watching the beam of a lighthouse revolve about 120 degrees in from of the speakers.

And while in reality your speaker will not move, the frequency dependent nature of comb filtering 'will'! And thus broad band feeds will result in the superposition of the various polar responses for the different frequencies. There is ONE way to double the gain and not cause the variations in comb filtering and polar patterns of the single unit driver But that will jave to wait for another time. And if you are interested you can PM me, as they are not generally employed in small scale listening environments.

So I guess I have provided a small puzzle and a few ideas for thought. But this is a very dramatic demonstration of comb filtering, and polar response, especially when the impulse response, frequency response and polar patterns can be generated and displayed!

* If one wants to atempt to EQ the minimum phase components of the composite signals, may I suggest they take a look at the SIA_SMAART analyzer.

** (think Auratones or Bose satellites - and you were wondering what they were useful for aside from their use as mediocre paperweights!)

*** (preferably NOT from separate amp channels! As these results are often surprising, and the results likewise less spectacular due to differences in phase between channels!)

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